i'm trying to build iOS native audio/video chat and stucked on audio. The sound is lagging, missing some parts and distorting. I tried this with r9919 and the latest one (r10184) builded by pristine build scripts. But when I tried to use older versions (r8444, r8926, r9132 and r9137) taken from PerchRTC Demo project (https://github.com/perchco/perchrtc - fat lib and public headers) everything seems ok (except little echo). What preactions should I perform to achieve at least the same streaming quality as (r8444, r8926, r9132 and r9137) provide? I also tried to use different audio codecs and different media constraints, still have no luck. I create audio connection this way
RTCMediaStream *localStream = [_pcFactory mediaStreamWithLabel:@"ARDAMS"];
RTCAudioTrack *localAudioTrack = [_pcFactory audioTrackWithID:@"ARDAMSa0"];
localAudioTrack.delegate = self;
[localStream addAudioTrack:localAudioTrack];
[self.peerConnection addStream:localStream];
[self.peerConnection createOfferWithDelegate:self constraints:_constraints];
using this constraints (tried different combinations)
mandatoryConstraints = @[
[[RTCPair alloc] initWithKey:@"OfferToReceiveAudio" value:@"true"]
];
optionalConstraints = @[
[[RTCPair alloc] initWithKey:@"internalSctpDataChannels" value:@"true"],
[[RTCPair alloc] initWithKey:@"DtlsSrtpKeyAgreement" value:@"true"]
];
Also I tried to different manipulations with AVAudioSession before obtaining audioTrack with no luck:
AVAudioSession *audioSession = [AVAudioSession sharedInstance];
[audioSession setCategory:AVAudioSessionCategoryPlayAndRecord error:&categoryError];
[audioSession setMode:AVAudioSessionModeVoiceChat error:&modeError];
[audioSession overrideOutputAudioPort:AVAudioSessionPortOverrideNone error:&overrideError];
[audioSession setActive:YES withOptions:AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation error:&activeError];
SDP descriptions look similar in lagging and not lagging builds (this one is offer - a=setup:actpass,answer has a=setup:active):
sdp = "v=0
o=- 7772121714021031999 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS ARDAMS
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 127 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:XLpFvB+JEaSN7tww
a=ice-pwd:9hoMfb7AJ9jC6Weej7qqTWkT
a=fingerprint:sha-256 AE:73:33:DD:31:CA:84:5A:96:4D:68:27:A0:23:82:3C:08:3B:7F:7B:A2:FE:91:1D:A7:3A:1F:2A:58:4B:FF:A2
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:127 red/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2797474154 cname:43zbAmj6VvYHT31F
a=ssrc:2797474154 msid:ARDAMS ARDAMSa0
a=ssrc:2797474154 mslabel:ARDAMS
a=ssrc:2797474154 label:ARDAMSa0
Any suggestions?
The Demo is using the top speaker by default. You need set to main speaker using code like this:
AVAudioSessionPortOverride override = AVAudioSessionPortOverrideSpeaker;
[RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeAudioSession
block:^{
RTCAudioSession *session = [RTCAudioSession sharedInstance];
[session lockForConfiguration];
NSError *error = nil;
if ([session overrideOutputAudioPort:override error:&error]) {
_portOverride = override;
} else {
RTCLogError(@"Error overriding output port: %@",
error.localizedDescription);
}
[session unlockForConfiguration];
}];